1. Field of the Invention
The present invention relates generally to home networking. More specifically, the present invention relates to managing telephony services using multiple users within a telephony control point in a home network.
2. Description of the Related Art
Home networking has advanced from the early days of merely linking computers and printer to the modern home network, which can include mobile devices, televisions, set-top boxes, refrigerators, etc.
Universal Plug and Play (UPnP) is a distributed, open networking architecture that allows devices to connect seamlessly and to simplify the implementation of networks in the home (data sharing, communications, and entertainment) and corporate environments. UPnP achieves this by defining and publishing UPnP device control protocols built upon open, Internet-based communication standards.
UPnP has grown in popularity of late in part due to the rise in popularity of media servers. Media servers are small computers that store multiple types of content (e.g., photos, music, videos, etc.). The content may then be streamed from a media server to one or more control points (e.g., iPod, television set, etc.).
Voice over Internet Protocol (VoIP) is a general term for a family of transmission technologies for delivery of voice communications over the Internet or other packet-switched networks. Other terms frequently encountered and synonymous with VoIP are IP telephony and Internet telephony, as well as voice over broadband, broadband telephony, and broadband phone, when the network connectivity is available over broadband Internet access.
VoIP systems usually interface with the traditional public switched telephone network (PSTN) to allow for transparent phone communications worldwide.
VoIP can be a benefit for reducing communication and infrastructure costs by routing phone calls over existing data networks and avoiding duplicate network systems. Skype™ and Vonage™ are notable service provider examples that have achieved widespread user and customer acceptance and market penetration.
Voice-over-IP systems carry telephony speech as digital audio, typically reduced in data rate using speech data compression techniques, packetized in small units of typically tens of milliseconds of speech, and encapsulated in a packet stream over IP.
The Session Initiation Protocol (SIP) is a VoIP signaling protocol, widely used for setting up and tearing down multimedia communication sessions such as voice and video calls over the Internet. The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. The modification can involve changing addresses or ports, inviting more participants, adding or deleting media streams, etc.
SIP clients typically use TCP or UDP (typically on port 5060 and/or 5061) to connect to SIP servers and other SIP endpoints. SIP is primarily used in setting up and tearing down voice or video calls.
A motivating goal for SIP was to provide a signaling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the public switched telephone network (PSTN). SIP by itself does not define these features; rather, its focus is call-setup and signaling. However, it was designed to enable the construction of functionalities of network elements designated Proxy Servers and User Agents. These are features that permit familiar telephone-like operations: dialing a number, causing a phone to ring, hearing ringback tones or a busy signal. Implementation and terminology are different in the SIP world but to the end-user, the behavior is similar.
SIP-enabled telephony networks can also implement many of the more advanced call processing features present in Signaling System 7 (SS7), though the two protocols themselves are very different. SS7 is a centralized protocol, characterized by a complex central network architecture and dumb endpoints (traditional telephone handsets). SIP is a peer-to-peer protocol, thus it requires only a simple (and thus scalable) core network with intelligence distributed to the network edge, embedded in endpoints (terminating devices built in either hardware or software). SIP features are implemented in the communicating endpoints (i.e. at the edge of the network) contrary to traditional SS7 features, which are implemented in the network.
Recently, VoIP has been extended to mobile devices such as cellular phones. There are several methodologies by which a mobile handset can be integrated into a VoIP network. One implementation turns the mobile device into a standard SIP client, which then uses a data network to send and receive SIP messaging, and to send and receive RTP for the voice path. This methodology of turning a mobile handset into a standard SIP client requires that the mobile handset support, at minimum, high speed IP communications. In this application, standard VoIP protocols (typically SIP) can be used over any broadband IP-capable wireless network connection.
As UPnP grows in popularity, more and more devices in the home are going to be networked. If these devices all have the capability to perform various telephony-related tasks through the UPnP protocol, then it is desirable to make the telephony aspects of these devices as easy to use as possible. Additionally, while traditional telephones were typically only used by a single user (or at least, only a single user at a time), many UPnP devices may be more communal-type devices, such as televisions and refrigerators, where it is more likely there may be more than one user operating the device at any one time. For example, a phone call may arrive for Dad at a UPnP television, but Dad is watching the television with Mom and the kids as well. Indeed, with the presence of multiple UPnP devices in a home network, it may be that Dad is closest to one device while Mom is closest to another. It would be beneficial if there was a way to extend telephony services so that it is tied to individual users such that a call or request for presence information would be directed only to the most appropriate UPnP device. The current UPnP standard only permits tying telephony services to control points, not users.